[ previous ] [ next ] [ threads ]
 To :  Diana Cionoiu <diana-liste@v...>
 From :  Meftah Tayeb <tayeb.meftah@g...>
 Subject :  Re: [yate] sip <-> sip-t signaling proxy
 Date :  Thu, 03 Jun 2010 20:53:34 +0200
hi diana,
sory, i was having problem finding a solution that do sip-T and i didn't 
know that yate do it!
thank you for the info and while hack YATE a litle more to understand it

Le 02/06/2010 11:58, Diana Cionoiu a écrit :
> Hello Meftah,
>
> I'm sorry to disapoint you but Yate supports SIP-T.
> Because Yate has his own ISUP implementation done in the right way, we 
> can use it together with the SIP implementation. Therefor we do have 
> SIP-T.
>
> Diana
>
> Meftah Tayeb wrote:
>> hi,
>> i think no
>> no yate, asterisk, openser, opensips, freeswitch or evean cisco don't 
>> support sip-T
>> go do cleare/standard sip
>> Le 01/06/2010 17:16, Alexandr Gevlichenko a e'crit :
>>> Hi all
>>> I need to configure sip <-> sip-t proxy with some unusual features:
>>> 1. Ability to implement in IAM (Attached to the INVITE) message 
>>> "redirecting number".
>>> For example, if the initial INVITE message contains an additional 
>>> header "Redirecting number" must be added to the IAM message.
>>> 2. Normal work as a sip <-> sip-t signaling proxy. It is planned to 
>>> be used in the scheme ASTERISK - (sip) - YATE - (sip-t) - PSTN.
>>>
>>> YATE could be this proxy?
>>> Thanks. 
>>
>


-- 
Meftah Tayeb
algérie télécom SPA
phone: +21321761805
phone (INUM): +883510001289101
mobile : +213660347746
mobile (INUM: +883510001289110
http://www.algerietelecom.dz